Added support for using VST plug-ins for audio effects processing.
This is integrated into the same page as the DX-plug-ins.
In the "Effect setup" dialog (which shows the VST plug-ins GUI window), you can load and save plug-in preset files from the system menu (click on the small upper-left window icon of the dialog).
In the "Format options" dialog, you can set one or more VST search path(s) (; separated - if left blank, it will retrieve the standard Steinberg path from the registry).

Added a new "channel format" option named "Split files; 0 1 ...". This splits multi-channel input files into multiple mono output files, adding the channel index to the file name (e.g. "x.0.wav", "x.1.wav" et c).
Also, the old "Dual files 'L+R'" option has been replaced by "Split files; l r ..." which works similarly, but adds one or more letters indicating the speaker placement instead the index (e.g. "x.l.wav", "x.r.wav", "x.lfe.wav").

Added an "<Auto select>" dithering option.
This selects "None (round to nearest)" if the output bit depth is ≥ the input bit depth and no sample modifying audio processing is done (resampling, plug-ins, et c).
In all other cases it selects "Noise-shaped dither".
NB; Like in previous versions, the dithering option is disabled if the output data format is floating point or use a compression codec that accepts floating point data.

The conversion progress is now also shown in the taskbar (Windows 7 / 8 only).

For writing BWF .WAV files (Broadcast Wave Files), the "Write EBU 'levl' chunk" option (in the "Format options" dialog) has been complemented by a block size selection and a choice of storing either only "Absolute peaks" or both "Pos. + Neg." peak values.

The output file format list is now formatted so that you can type in the first letter of the file extension to search in the list.

Version 11.0

Most file formats now have a new "<Auto select>" data format selection.
Select this to let the file writer decide what data format to use.
It does this by querying the file reader about the input file's data format - if it is a data format that the writer can also handle, then it will use that (thus preserving the data format of the input file).
Otherwise, it will select a default data format (e.g. "PCM 16-bit").

Added "Direct Stream Copy" support for MPEG audio layer II, MPEG audio layer III and MPEG AAC compressed data (a.k.a. MP2, MP3 and AAC).
NB, you *must* set the output data format to "<Auto select>" for this to work, and it will only work for file format for which the program normally supports writing the resp. data format.
And just like "Direct Stream Copy" with uncompressed formats, it can only be used if the audio data is not modified in any way (so no resampling et c).
When it can be used, it has the advantage of copying the compressed stream verbatim (e.g. from an .AVI file to a .MP3 file) without loosing audio quality due to recompression.

When the "Normalize" feature is set to "output meta data" (but not when set to "modify audio") it will now support "Direct Stream Copy" for MPEG audio layer II, MPEG audio layer III and MPEG AAC audio (in addition to uncompressed formats).
The practical upside of this is that you can run MP2, MP3 or AAC data through the program and calculate gain adjustment (e.g. ReplayGain) which is added as meta data to the output file — without degrading the compressed audio.

Added support for reading and writing raw AC3 audio streams (.AC3), raw padded DTS audio streams (.DTS), and raw compact DTS audio streams (.CPT).
Please note however, that the program does not contain any AC3 codec so by itself it can neither compress nor decompress these types of data.
However, if you have installed a "Windows ACM filter" that can decode AC3 or DTS (e.g. the common "AC3Filter"), then it can use that to decompress such data.
There's currently no way for the program to compress data to these formats.
What you can now do though, is to copy compressed streams between files using new "Direct Stream Copy" support for AC3 and DTS.
File formats that supports this are: .AC3, .AVI, .CPT, .DTS, .MKV, .MOV, .WAV (NB: .AVI, .MKV and .MOV are read only, and for the others you must select "AC3" or "<Auto select>" as output data format for the copy to work).

Added support for normalization per EBU R 128 (this is basically the ITU BS 1770 Leq(R2LB) loudness measure + two gating functions + definition of a "0 LU" reference level).
NB, the EBU R128 target level of "0 LU = -23 LUFS" lies at approx. -5dB compared to the Replay Gain target.
So if you wish to test to use EBU R 128 instead of ReplayGain, then you may want to enter a target value of 5 LU to compensate.

The normalization options "-20-Leq(RLB)" and "-20-Leq(R2LB)" have been replaced by "-Leq(RLB)" and "-Leq(R2LB)", with a default target level of -20 dBFS. These algorithms, both from ITU BS 1770, will now also work at sample rates other than 48KHz.

When normalizing the audio volume using the Replay Gain methods, the target value box now allows you select the desired reference level in dB(SPL). The original Replay Gain document specifies calibration against a 83 dB(SPL) reference level, but the majority of software today use 89 dB(SPL) instead (because the original value was deemed to be too low).

Added a normalization option to find the "True Peak Level". Whenever you enable any of the normalization types, the peak sample value is also determined, and is saved as meta data (if the output file format supports it).
With true peak enabled, it will also examine the signal at time points between the original sample values (using a 16x oversampling filter - for better precision than the 4x demanded by EBU R 128).
This comes closer to the true peak that a DAC will have to handle.

The channel icons in the input file list now better corresponds to actual the speaker layout (if known), not just the number of channels.

The "Mixing" tab of the file options dialog now allows you to indicate which speakers are be used (this info can be saved in .MOV, .W64, the "Microsoft extensible" version of .WAV, .WMA, and .WV.).

The "Format options" dialog box now allows you to select the sample rate for Rockwell ADPCM files (typically either 7200 or 8000 Hz).

For writing MPEG layer II compressed data, a new v1.3 of tooLameF.dll is required (available from our web-site).

Various minor file format-related improvements.

Version 10.5

The resampling speed is much improved - changing the sample rate is now up to 9x faster than before.

Conversions are now often faster for the case when no audio processing is needed (i.e. no resampling et c), and the input and output file use identical data and channel formats. This is due to support for a new "Direct Stream Copy" path in the underlying awC++ library (NB; this is currently only available for uncompressed data formats).

Updated the codec add-on pack (now v1.7) with the latest versions of various external dll's.

Version 10.4

Added a "Play selected" option to the "Add files" dialog where, when you select a file, you can hear its contents playing after 1s. The dialog is now also resizeable.

Added support for the Windows Audio Sessions API (WASAPI) for audio playback. This is the "native" audio interface for Windows 7 and Vista, providing low-latency, high quality audio playback (when running on Windows XP, the DirectSound API is used instead).

Added support for reading and writing Musepack (.MPC) compressed files (SV7 and SV8 types supported for reading, SV8 only for writing).

Added support for reading and writing files with Shorten (.SHN) lossless compression.

Several improvements to the normalization function:

Fixed a conformance problem with Replay Gain meta data calculations (if you have used an earlier version to calculate Replay Gain meta data for files, then it is recommended that you use a "tag editor" of your own choice to rescan them and calculate new Replay Gain data). This selection is now called "Replay Gain 'Standard'".

The new selection "Replay Gain 'ISO 226:2003'" adds a fairly accurate filter implementation based on the "75-phon equal loudness contour" from the "new" research standardized as ISO 226:2003. This is an alternative to the standard Replay Gain hearing filter (which is an approximation of the "80-dB F-weight curve", based on research as old as 1933!).

Also added new "Leq(RLB)" and "Leq(R2LB)" selections for loudness measurement (the latter is also know as ITU BS 1770). NB, these are currently only supported for 48KHz.

In some informal tests that we've done, on music-only material Replay Gain with the ISO-based filter performed 14% better than Replay gain with the standard filter, and 18% better than Leq(R2LB).
For speech material though, Leq(R2LB) outperformed both Replay Gain variants by 20-25%, although it should also be noted that the number of sound clips used for the speech tests were fairly small, so results may not hold up to closer statistical analysis.

Several improvements to the "Dithering" option (called "Quantization" in previous versions):

The selection box is now disabled when selecting a data format where no quantization has to be made (e.g. "Float 32-bit") or where the encoder accepts the data in floating point format and does its own quantization if necessary (e.g. .AAC, .MP2, .MP3, .MPC, .OGG).

The dithering selection is now stored in preset files.

Added Unicode support for file names.

Added Unicode support for text meta data (for file formats that support it). When saving to meta data formats (e.g. ID3v2 tags) that supports multiple character encodings, the program will now automatically select the most compact one (from Latin-1, UTF-8 or UTF-16) that can represent the text without losing any data.

Preset files (.aap) are now stored in app-data folder (they were previously stored in the program folder, where Vista UAC didn't like them to be created).

Added support for reading text meda data in .WAV files from a "cart chunk" used by some radio broadcasters.

Added support for encoding 12 or 20-bits/sample FLAC files (in addition to 8, 16 or 24-bits/sample)

Added selections for iPhone ringtones (.m4r, same thing as the .m4a format).

Added ASIO support to the audio recording wizard.

Version 10.1

Added support for reading and writing Multichannel Broadcast wave format (.WAV) a.k.a. EBU RF64. This is an extension to the Broadcast wave format (which is in turn based on the normal Microsoft wave format). It adds features to support for file sizes > 4 GB, as well as for high bit-depth and multi-channel audio.

Added support for reading and writing WavPack (.WV) lossless compressed files. NB; to use this you need to install the free WavPack codec add-on (wavpackdll.dll).

The program now supports handling of input files containing more than one 'audio stream'.
Currently this is used only with .AVI and .MOV (where each 'audio track', if there's more than one, shows up as a separate entry in the input files list), and with .VAP (where each voice prompt appears as a separate entry after opening a file).

It is now possible to continue working with the input file list while at the same time keeping an Audio Player window open.

Added support for writing ID3v2 tags to .MP3, .MP2 and .AAC files (disabled by default - you can enable it in the 'Format options...' dialog - formerly known as the 'More options...' dialog).

Added support for an alternative way to store ReplayGain info in ID3v2 tags.

A note is now written to the log file if log + normalization + 'never clip' options are all enabled and the volume had to be lowered in order to avoid clipping.

If you are running under Windows XP then buttons and other control will now use the rounder 'XP style' look.

The program manual is now in 'HTML help' format (it now both looks better and is easier to use than when it was using the older 'WinHelp' format).

Version 9.3

The "More info..." dialog has been replaced with a "File options" dialog, containing three new sub-pages: "File info", "Metadata", and "Mixing". To open this dialog, simply double-click on an entry in the input files files, or right click on one and select "File options...".

The new "File info" page contains a list of information about the format and contents source file. Additionally, it contains two new "per file" options: A "File name override" option allows you to give the output file (i.e. after conversion) a different name than the input file. A "Sample rate override" option allows you to override the sample rate derived from the files (this does not change the audio data - it is sometimes useful for doing a manual correction e.g. if the input file format did not contain any sample rate information).

The new "Metadata" page not only allows you to view all metadata collected from the source file, it also allows you to edit and override it (note, this affects the meta data that is passed to the output file only, the input file is not touched in any way!).

The new "Mixer" page allows you to arbitrarily map up to 10 input channels (from as many input files) into the output channels of your choice, and with volume and time offsets adjustments for each if you want. The old "Combine two mono files" command is now implemented as a quick way to set up this mixer for the special case of merging to mono files into one stereo file.

Output files are now written first to a temporary file name that is then renamed to the final output file name after the conversion is complete - if and only if the conversion succeeded - if the conversion failed for whatever reason, then the temporary file is deleted. Note, combined with the "Delete source files" options it is not possible to replace an original file with a new one using file name & extension.

Added support for reading & writing "MPEG-2 and MPEG-4 Advanced Audio Coding" format (.AAC files) by using an external 3rd party encoder called FAAC (using 'libfaac.dll') and decoder called FAAD 2 (using 'libfaad2.dll'). These are open source software projects and should be generally available on the net - however they are not available from us. ADTS, ADIF and 'raw' container formats are supported for reading, ADTS only for writing - see the help file under file formats for more information.

Added support for ".U255LAW" files. This is an 8-bit "exponential" data format used by old drum computers. There is not standard extension for these files, and no way to auto-detect them, so you need to use .u255law to get the program to recognize them. Also, there's no way of knowing the sample rate - the program will assume 24000 Hz when reading these files.

Added decoding of Yamaha 4-bit ADPCM data type from .WAV files as well as both reading and writing it in 'raw format' as .YADPCM files.

Increased the maximum number of simultaneous output formats from 5 to 10 (by adding a second "multiple output" page).

Updated the APE codec to v3.99. This should solve an issued with some newer .APE files not opening.

Version 9.2

Added support for reading ID3v2 meta-info extension in .AIFF files.

Added 'de-emphasis' support for reading audio tracks (.CDA) from some rare (mostly older) Audio CD's. Very few audio CD-rippers supports this, but without it those rare CD's recorded with 'emphasis' will sound wrong!

When writing "Broadcast Wave Format" files, the correct extension .WAV is now used (it is now called "Microsoft Waveform - BWF 'Broadcast' type").

Improved Wave-64 (.W64) support (many more data sub-formats are now supported both for reading and writing).

Added a "Never raise to clipping" option to the normalization function. Enabling this will put a cap on how much the normalization algorithm may raise the volume ensuring that it is never raised so much that clipping occurs.

Added detection of special LAME and Xing 'tags' inside MP3 files. From these tags encoder delay and padding information is extracted. This means that the short 'silent delay' that is usually present at the start of a decoded MP3 file can now be removed for those files.

Added a new output channel format specification: "Dual files 'L+R'". Selecting this will create two mono files as output, one containing the left channel data, and one with the right channel data.

The program now auto-detects "dual mono file pairs" when you add a file - i.e. if the there exists two mono files with file names ending with " L", " R", "-L", or "-R" and you add one of them. When this is found, you are asked you if you want to combine them (you can answer 'Yes', 'No', 'Yes - Always', or 'No - Never'). Also, when using this, or (manually using the "Combine two mono to stereo" function), the trailing letter is automatically removed from the output file name.

Added support for reading and writing Soundscape Audio-Take files (.ATAK1A through .ATAK4D extensions).

There are now two FLAC output sub-formats: "Encoder default settings" and "Maximum compression". The latter is what previous versions always did, but this had the side effect of quite long compression times. Using the new "Encoder default settings" gives a bit less compression, but is much faster and should provide equal performance as most other FLAC compression software.

Added a "Show more info" command that you reach by right-clicking on a file in the input files list.

Added an option to "Write EBU 'levl' chunk" for BWF files (see the "More options" dialog). This pre-computes and stores a list of peak-level information in the BWF file which enables faster loading and display in some applications.

Added a "Use more built-in effects" option and moved normalize, trim length et c to their own page. Also moved the bit-depth quantization options from the "More options" dialog to the output options page.

The "Next >" button before the progress page has been renamed into "Start >".

Moved all of the 'built in' audio processing options to their own page called 'effect options' (except 'resampling' which is so common that it is included both in this new page, and on the main 'output options' page).

Added new "Jitter correction" and "Scratch correction" options for CDDA-extraction (found in the "More options..." dialog). Also added a "Read cmd" selection, where you can select alternative CDDA read modes in case the default mode does not work.

Fixed a few minor bugs and issues.

Version 9.0

Added two more normalization methods: 'Replay Gain - Modify audio data' and 'Replay Gain - Store meta info in file'. These methods use the excellent 'Replay Gain' algorithm for audio normalization. With these, there's now also an 'Album gain mode' checkbox option - enabling it will calculate the normalization gain over all the input files instead of 'file by file'. For more info on Replay Gain, see http://replaygain.hydrogenaudio.org/. Note that the gain value that can be entered in the options for 'Modify audio data' is the desired deviation from the Replay Gain standard value of 83 dB SPL (typing in 0.0 means that you want it at '83 dB SPL' when played back on movie industry standard listening consoles). When using the 'Store meta info in file', the deviation from 83 dB SPL is stored in file formats that support it (see below), and the actual playback level is then supposed to be controlled by a Replay Gain compatible player.

Added 3 new options for noise-shaped dithering: 64/128/256 point F-weighted noise shaping filters.
The higher the number of filter points, the more accurately is the noise shaped to the 'inverse' of the ears sensitivity, but also the more computationally expensive... These new options should give you results close to the best specialized professional dithering products!

The audio quality of the FIR resampling algorithm has been improved quite a lot by increasing the precision of the filter calculations. The previous two 'FIR (13-tap)' and 'FIR (65-tap)' selections has been replaced with five new: 'FIR (dirty - very fast)', 'FIR (decent - fast)', 'FIR (good - normal)', 'FIR (super -slow)' and 'FIR (extreme - very slow)'. The first two of these approximately corresponds to what was available in previous versions. The last three take quite a lot of CPU power to compute, but also gives very, very good results! The improvement of the last two of them is only really measurable 20 or 24-bit data though.

Added a "Combine two mono to stereo output" function. To use it, right click, on one or more mono files in the input file list, then select the command from the context menu. There you may also find a "Clear combine" command that undoes this setting.

Added .FLAC format read and write support. This format provides good loss-less audio compression. For more info (including player plug-ins), see http://flac.sourceforge.net/. To use this you need to download the free libFLAC.dll add-on from our web site and copy it to the directory where you installed the program.

Added support for reading and writing raw ITU G.728 LD-CELP data (.G728). These files do not have any 'header info', which means that you need to use a file extension of .G728 for the program to be able to recognize them. Also, before reading these files, be sure to go into the 'More options...' dialog and select if 'codeword padding' should be used, and if the data is stored in big-endian or little-endian format.

Added support for a few more meta-data types (composer, publisher and sub-title).

Added support for reading & writing 'cue points' in WAV files.

Added read & write support for 'APE tags' containing meta info for .APE and .MP3 files. For MP3 files this info is only written if it is enabled in the 'More options' dialog.

Added limited support for reading ID3V2 meta data (.MP1/.MP2/.MP3 files). Compressed and/or encrypted meta-data is ignored. Only the 11 most common meta-data types are handled. ID3V2 v2, v3 and v4 is supported.

Added support for gain adjustment values stored in WAV, AIFF, IFF, APE, OGG, FLAC and MP3 files.
In WAV files it is stored as 'Replay Gain' info in an 'rgad' chunk. This is written only if the input file had Replay Gain info, or if the 'Normalize' option is enabled and the normalize type 'Replay Gain - Store in file' is selected.
In MP3 files it is stored as 'Replay Gain' info either in 'ID3v2 format', or in 'APE tag' format. The former is currently only supported for reading, while the latter is supported for writing but you must enable it in the 'More options' dialog.

You can now also create MP3 files with 8000, 11025 or 12000 Hz sample rate (previous versions used 16000 Hz as a minimum sample rate). This requires v3.94 or later of LAME_Enc.dll.

The 'More options' dialog now has options for writing joint stereo MP2 and MP3 files. This was always on in earlier versions, now you can deselect it if you prefer the 'normal' MPEG stereo mode. The 'joint stereo' mode usually gives better compression performance, but some claim it can damage the important phase-coherence in Dolby ProLogic encoded material. For MP2 data, you can now also select if you wish peak energy levels to be stored in the 'frame aux-field' (this is required by e.g. Digigram pcx cards). Note, for using this with MP2-files, you need to download a new compile of TooLameF.dll (there's a link to it on the Awave Audio download page on our website).

For the silence removal option you can now select the minimum length of sections to be removes (sections longer than this is ignored).

It is now possible to use the limiter function even when not using the normalize function.

When doing normalization operations, the status text in the 'Processing batch' dialog will now indicate first 'Analyzing' and later 'Converting'.

When a batch conversion is finished, the 'Cancel' button now changes name to 'Close', which works like the 'Finish' button previously did. The latter has been replaced by a 'New batch' button, which now brings you back to the first wizard page.

Version 8.3

The DirectX plug-in effect settings have now been moved to a separate page (check 'Use DirectX plug-in effects' on the main options page, then 'Next' to get to these). Plus, you can now use up to 6 effects.

Added a 'Limiter' option. Use this together with the normalize option ensure against clipping when normalizing (when the target level is set too high).

Added the option 'Skip existing files without prompting'.

Added an elapsed time display on the progress page.

Added a 'Create log file' option that writes an 'Awave Audio log.txt' file to the directory where you installed the program.

Added an option called 'Multiple output formats'. Checking it will give you an extra page (after the 'Select output options' page), where you can select up to 5 different output formats to be produced in the same batch-run.

You can now select the 'quantization method' that is used when down-converting the sample bit-depth (e.g. input 16-bit PCM → output 8-bit PCM). Available selections are:

Round to nearest. This is the fastest and most commonly used method.

White noise dithering. White noise is added below the least significant bit (lsb) and errors are carried to the next sample. This is a minor improvement of the method used in previous version of Awave Audio.

Noise-shaped dithering. 9th order shaped (F-weighted) noise is added and errors are carried to the next sample. This is the option with the best perceptual audio quality when down-converting the bit-depth. But note that when you are not down-converting, you will not get bit perfect copies due to the level of added noise being above the lsb.

Takes 32-bit floating point data as input. This means that the audio data no longer have to be rounded off and dithered before it is passed to the encoder - the encoder's psycho-acoustic model will then determine what to throw away. Regardless of if the original data is 8-bits/sample or 24-bits/sample; you'll get the best result possible for the selected bit-rate.

Some quality optimizations.

Fixed several memory leaks.

The new DLL file is available on our web site in the archive toolamef02l.zip

Added support for reading and writing Monkey Audio losslessly compressed files (.APE). This format is well suited for compressing important archive data without loosing any quality - for more info (including player plug-ins), see http://www.monkeysaudio.com/.

Added G.721/723/726 ADPCM sub-format write support for .AU files.

Added support for adding 'ID3 tags' when writing MP2 files. NB, you need to enable this in the 'More options' dialog if you want it.

The channel format selections 'Left channel' and 'Right channel' has been replaced with 8 selections named 'Channel n' (where n is 1..8) allowing you to pick out any one of up to 8 channels from the input data.

The 'normalize' function now buffers the data on disk in the 'first pass' (it is a 'two pass' algorithm) so that it can run faster in the 'second pass' (previously the input file had to be re-decoded a second time).

Improved the Rockwell ADPCM codec.

Changed the order in which DirectShow effects are processed (they previously processed the audio data in reverse order to the plug-in numbering, which could be a bit misleading...).

Version 8.1

Added support for writing the new Windows Media 9 Format files (this requires the Windows Media 9 runtime to be installed - available from the download page on our web site!). This includes support for 2-pass CBR (constant bit-rate) as well as 1-pass VBR (quality based variable bit-rate) and 2-pass constrained VBR (a.k.a. 'average bit-rate' encoding). You can also use external 'profiles' (.prx files) for things such as MBR (mutual bit-rate exclusion) and other kinds of multi-stream files.
All the current codecs are supported: Windows Media Audio 9, Windows Media Audio 9 Professional, Windows Media Audio 9 Lossless, Windows Media Audio 9 Voice, and ACELP.net. Plus, there's support for writing older Windows Media Audio 4 format WMA files.
NB; Windows Media Audio 9 Professional multi-channel audio is only supported on Windows XP (and later).

Added support for reading the audio portion of any ASF and WMV file (also requires the Windows Media 9 runtime).

Added '5.1/6.1/7.1 surround' selections to the output channel format list for many formats.

The icon to the left of files in the input file list now indicates 1, 2, 3, 4, 5, 5.1, or 6.1+ channels of audio.

Files with extension .ALW and .ULW are now recognized as raw A-Law and mu-Law format files.

Added a 'Fade' option to the 'trim length' function - which performs a 250ms fade out at the trim point.

Added support for writing NIST SPHERE files (Usually .NIST).

The previous "Normalize RMS" option has been renamed "Normalize RMS of Amplitudes" and a new "Normalize RMS of the Power" option has been added.

Added detection of .VAG files with missing 'VAGp' header id.

It is now possible to pass preset files (.aap) on the command line.

Added a context menu when you click on a file in the input file list.

Fixed a bug where writing .GSM formats did not work.

Fixed another bug with loop points not being adjusted correctly when audio is resampled.

Version 8.0

Rewrote much of the internal file format handling in order to improve how 'sub-formats' are handled. This is mostly noticeable in that formats that support many output bit-rates are no longer listed as multiple file formats (as they had to be in previous versions). Instead they are listed as a single format - but with multiple bit-rates available in the sub-format list. Also many sub-formats now have more descriptive names.

A new option 'Keep the input file sub-directory structure' let's you copy a desired level of sub-directory names from the input file path to the output file path. It can be specified either as the number of lower levels to keep, or as the number of upper levels to discard. E.g. if you select to keep the 2 lower levels and have the input file 'c:\foo\bar\sounds\myfile.wav' and the output directory 'C:\out' then the output file will be written to 'c:\out\bar\sounds\myfile.au'.

Now checks if a file is already in the input list when adding one (and does not add it again if it was).

It now supports SPTI (SCSI Pass Through Interface) for reading Audio CD's on Windows NT/2K/XP in addition to ASPI32. This means that ASPI32 drivers are no longer needed under those Windows versions. You need to be logged in with administrator privileges to use SPTI though.

Improved support for writing MP3's (using Lame_Enc.dll) - it now not only supports writing 'Constant Bit Rate' .MP3 files, but also 'Average Bit Rate' and 'Variable Bit Rate' files:

ABR is really a form of VBR where you specify the total size of the output file (indirectly, since size = length * bit rate). The ABR encoder then 'portions out' the total amount of available bits in variable amounts to different parts of the audio clip in a manner that it thinks will produce the best audio quality.

The VBR encoder on the other hand let's you specify a 'quality level' and then varies the bit rate so as to always achieve at least that given quality, without any concerns about what the total file size may end up being. This is the optimal way to compress the audio!

Note that not all MP3 decoders supports ABR and VBR files... Also, you need the latest Lame_Enc.dll (version 3.92 or later) to use this (On the Awave Audio download page, on our web site, there's also a link to where you can download this).

Added an option dialog to the 'Add dir' command and the possibility to specify one ore more filter strings (with wild-card support, e.g. '*.wav').

The 'Input files' list now have separate columns for 'File name' and 'File path'.

The final release version 1.0 of the Ogg Vorbis codec (.OGG files) is now available - get the separate add-on (vorbis10.zip) from our web site. You now have a larger selection of bit-rates, including low bit-rate modes.

Added support for writing MPEG audio layer III sub-format .WAV files.

Added support for writing 4-bit DVI ADPCM sub-format .WAV files.

Added an 'Only at beginning' option to the 'Remove silent sections' function.

For the 'Normalize' and 'Remove silence' functions you can now specify the values in either dB or in percent.

Added an 'Exit when done' button to the progress page.

Added a 'Hide' button to the progress page.

You can now double-click on a file in the input file list to play it.

Added support for converting fractional loop point and fine tuning information.

Various minor fixes and improvements.

Version 7.6

Added support for using the Windows ACM (Audio Compression Manager) to decompress any sub-formats of .WAV files that are not handled internally (e.g. Truespeech, G.723.1, Voxware, ACELP, and Windows Media sub-formats).

Added support for writing MPEG layer II audio files (.MP2) using the free TooLame encoder. This is available on our web site as a separate add-on DLL.

Added 'Left channel' and 'Right channel' selections in the channel format list to allow a single channel to be picked from a stereo file and save to a mono file. To 'split' a set of stereo files into two mono, simply run the batch twice, once with left and once with right channel selected, and save them to different output directories.

Added a 'Trim length' option.

Added a 'Remove DC offset' option.

Added a 'Recursively scan sub-directories?' question when using the 'Add dir' function.

Added support for reading audio CD tracks (.CDA). This require that you have an MMC2 compatible CD-ROM drive capable of accurately reading audio data digitally. You also have to have 'ASPI32' drivers on you computer (most Windows installations already have this). Note that this is an unsupported 'bonus feature'.

Added support for reading Olympus DSS files (.DSS). You must also have the Olympus DSS Player v3.x installed to be able to read these as it makes use of a DLL file from Olympus.

Added an 'Audio player' window.

Added a new F.I.R. filter (65-tap) resampling option for those of you who wont mind using up lot's and lot's of CPU cycles for the sake of a marginally better result than the 13-tap filter.

Added an option to select the number channels and the number of bits per sample in 'data fork only' SD2 files in the 'more options' dialog.

Added support for writing the new 'extensible format' type of WAV files (with better defined support for multi channel and high bit depth support).

Added support for writing 'Annotated speech' files (.VAP).

Version 7.1

Added support for writing RealAudio (.RA) files. To use it, you also have to install a whole bunch external DLL's from RealNetworks (you can download these as an 'Awave RA add-on' from our web site).

Added a 'presets' selection feature which let's you quickly load and save all settings. This includes the DirectShow filter setup. Plus, all settings from the last time you ran the program are now remembered (and saved as a preset named 'Last batch').

In the input dialog there's now an 'Scan dir' button which will scan a directory (and subdirectories) for audio files to add to the batch job. You can also drag and drop directories from the Windows Explorer to do the same thing.

Added audio recording functionality - try the 'Record' button in the input page. NB, this is to be considered an unsupported 'bonus feature' of this program.

Changed the quantization algorithm from simple rounding to noise dithering (e.g. when going from 24 to 16-bit, or from 16 to 8-bit).

You can now pass input file paths and directory paths (to be scanned for input files) on the command line.

Added a 'Delete source files' option.

Added support for writing ID3 v1 'info tags' to MP3 files. It is off by default but you can enable it in the 'options' dialog. Note that even when turned on, an ID3 tag will only be written if the source file actually has any of the info that can be written to it (name/title, artist, album, year, comment).

Added support for reading and writing 24-bit Sound Designer II files.

Added support for reading and writing Ensoniq PARIS files (.PAF).

You can now sort the input list by clicking on the column headers (click once more to sort in reverse order).

Fixed a bug with handling of file names with dots in them.

Improved support for DirectX plug-ins that changes the number of audio channels.

Fixed various minor bugs.

Version 7.0

The audio conversion code has been completely rewritten and is now based on our new "awC++ r2 SDK". This has a number of benefits:

Low memory consumption. You can now convert really big files without running out of memory or getting severe disk thrashing from the virtual memory manager.

Support for up to 32-bit floating point precision. This even goes beyond 24-bit PCM precision - great for those '2496' fans out there!

Support for multi-channel audio (i.e., files with more then two audio channels).

Better progress reporting.

Other new features are:

An 'Options' dialog where you can select such things as e.g. default sample rates and bit-ordering for headerless formats.

MPEG encoding is now done through external 'DLL format' encoders. Both 'LAME_Enc.dll' and 'BladeEnc.dll' are supported, plus you can now select different output 'formats' for different bit rates.

DirectShow 'effects' now work with 32-bit floating point precision (and only that, filters that only can handle 16-bit PCM format are no longer supported).

The DirectShow 'effects' now has a '+' button next to the selection lists. Clicking on it brings up the 'properties' dialog with the settings for the selected effect.

The 'Play' button now works a little differently (and now requires DirectX to work).

In addition to the old 'Normalize' peak values options, you can now also normalize the RMS (root of mean of squares) to a user defined value.

We've added a nice new logo, plus some slight 'texture' to the dialogs and buttons to make them look better.

Because of the complete rewrite, about two dozen file formats from the previous version has been scrapped. Either because they were so obscure and rarely used that we felt it wouldn't be worth the bother, or because they were so closely related to synthesizers and/or trackers that they fit better in the domain of our Awave Studio application.

Version 6.0

This is the first version of Awave Audio. However, it inherits most of its functionality from the 'Batch Conversion Wizard' of our earlier product called Awave.